pm said "I get good results by recording at a deliberately low level -- low enough that you're certain not to clip -- and then Normalizing later."

I haven't opened my DSP textbook in a while, but it seems to me that recording at a low level gives up a lot of data, sort of like sampling at a depth of 4 bits instead of 16. Normalizing that recording cannot recover the data that you gave up. Am I wrong?

I feel that unless you record at the max level possible without clipping, you are giving up part of the signal, no matter what you do with it afterward.

Isoprop usually has fewer impurities -- ones that will leave a residual
when the alcohol evaporates than most solvents. I would like the 1005
for this purpose because of less water.

And I think isoprop is also intoxicating, as is methanol, but both are
much more poisonous than ethanol. (remember the stories that one sees
occasionally about people buying an intoxicating drink and either dying
or going blind).

Chuck

On 9/3/12 4:02 PM, rfwilmut wrote:

Quote:

CDJonah_alt wrote:

Quote:

I don't think isoprop is usually denatured with anything the way ethanol is.

That's correct, because it's not drinkable. Meths is ethyl alcohol (the drinkable stuff) deliberately polluted with 10% methyl alcohol (poisonous), pyridine to make it stink and a purple dye to identify it. This renders the ethyl alcohol undrinkable (not that that stops some people) so it doesn't attract tax and can be sold for various purposes.

Isopropyl is not drinkable or intoxicating, so it can be sold as it is. It is less volatile than ethyl alcohol, so it's not quite as effective as a cleaning agent, but on the other hand it doesn't contain impurities. I'm not sure to what extent the impurities matter in this context (we used to use ordinary meths to clean quarter-inch tape heads in BBC recording channels) but with the very fine tolerances of cassettes isopropyl may be better.

That of course is true ... but ... remember the dynamic range of most
tapes is much less than 90 dB or so of most digitizers, so throwing away
10 dB really shouldn't matter much.

Chuck

On 9/3/12 5:19 PM, Bull wrote:

Quote:

pm said "I get good results by recording at a deliberately low level -- low enough that you're certain not to clip -- and then Normalizing later."

I haven't opened my DSP textbook in a while, but it seems to me that recording at a low level gives up a lot of data, sort of like sampling at a depth of 4 bits instead of 16. Normalizing that recording cannot recover the data that you gave up. Am I wrong?

I feel that unless you record at the max level possible without clipping, you are giving up part of the signal, no matter what you do with it afterward.

Thanks Chuck for your insight on recording at low levels and then normalizing. I played around with that for a bit and, at least qualitatively, think you're right.

How about that other useful trick (to avoid distorting the useful signal when suppressing hiss)? Is there any science behind this, or are we just suppressing 50% of the [some measure of the] hiss and distorting the signal less?

That hint was:

<quote>

A useful trick that's been suggested here in the past is to:

* Before sampling your noise selection, Amplify it by 50%

* Sample noise selection

* Restore amplitude of selection

* Suppress noise

The benefit of this is that it should halve the distorting effects of the
denoising process. If that doesn't succeed, you might want to try using some
different de-amplification ratios than 50, but 50 seems a good starting
point.

Quite honestly, no, I don't know of any science behind it, but then I am
not an expert in sound processing. But, in fact, I think I originally
posted that technique and only after I tried it on multiple examples and
found that it worked. I normally select a region and copy it to a new
file before doing the amplification.

Mathematically, what one is doing is getting an estimate of the
frequency distribution of the noise and then removing that frequency
distribution from that sample. I used to know the name of the filtering
but it has dropped out of my mind.

Chuck

On 9/4/12 12:52 PM, Bull wrote:

Quote:

Thanks Chuck for your insight on recording at low levels and then normalizing. I played around with that for a bit and, at least qualitatively, think you're right.

How about that other useful trick (to avoid distorting the useful signal when suppressing hiss)? Is there any science behind this, or are we just suppressing 50% of the [some measure of the] hiss and distorting the signal less?

That hint was:

<quote>

A useful trick that's been suggested here in the past is to:

* Before sampling your noise selection, Amplify it by 50%

* Sample noise selection

* Restore amplitude of selection

* Suppress noise

The benefit of this is that it should halve the distorting effects of the
denoising process. If that doesn't succeed, you might want to try using some
different de-amplification ratios than 50, but 50 seems a good starting
point.

If the noise level is, say, -40 dB and you increase it to, say, -20dB before sampling, then when you carry out the reduction process any wanted audio between -20 and -40 will be reduced in level on the assumption that it is noise, and the amount by which it is reduced will be different at different frequencies, so you are distorting wanted audio. The whole point of the sampling process is to determine below what level audio can be regarded as noise and therefore reduced in level, and to do so at each of a range of frequencies.

Thanks everyone for all your helpful suggestions.
Took delivery of 1litre of isopropyl alcohol, plus platene cleaner (to soften the rollers if they get hardened from cleaning) today.So on the right track..ha ha!
Just thinking...it would seem that background noise in redordings from small players like a Sony walkman is partly a function of having so many working parts in a small box, the Ion may suffer in the same way. I have a jack with which can connect the headphone outlet of my large portable Sony cd casette recorder to the line- in of a Bose cd radio. Then if i buy a line- out to usb cable i could connect that to the Mac and maybe that would produce a cleaner wave in Amadeus than using the Ion?

I tried to create a simple audio file to let me experiment with this issue, but what I got wasn't what I thought it was. I created a track 10 s worth of 400 Hz sine wave at 50%. I generated another track of white noise at 2%. I then merged the two tracks. Sure enough, it sounds like a nice tone with hiss overlaid, like a much-used cassette tape of a 400 Hz signal. I stuck a second's worth of 2% white noise on the front so I would have something to sample for suppressing noise.

I intended to try the trick of sampling the amplified noise before suppressing the noise in the whole file. But just to make sure I was on the right track I sampled the un-amplified noise and did the noise suppression with that. I expected the hiss (white noise) to vanish and something reasonably like a 400 Hz signal to remain.

Well, the whole wave form disappeared! Noise and sine wave were just gone, and there was nothing left.

Not surprising that I don't know how to interpret this result, since I don't know exactly what the Merge function does (although it appeared to do exactly what I wanted) and I don't know how AP's noise suppression works.

Any ideas what I did wrong, or suggestions on how I can create a simple file to experiment with this?

Then if i buy a line- out to usb cable i could connect that to the Mac and maybe that would produce a cleaner wave in Amadeus than using the Ion?

To connect a line-out to usb, you would also need an analog to digital converter box. Those cost from tens to hundrds of dollars. But you already have one built into the Ion. That's why it can connect via usb.

For converting audio cassettes, a simple AD converter should be good enough.

I don't think that is how the algorithm works, but as I say, I don't
remember the details. I will look it up this afternoon in my copy of
Numerical Recipes and get back on the subject.

Chuck

On 9/5/12 6:51 AM, rfwilmut wrote:

Quote:

If the noise level is, say, -40 dB and you increase it to, say, -20dB before sampling, then when you carry out the reduction process any wanted audio between -20 and -40 will be reduced in level on the assumption that it is noise, and the amount by which it is reduced will be different at different frequencies, so you are distorting wanted audio. The whole point of the sampling process is to determine below what level audio can be regarded as noise and therefore reduced in level, and to do so at each of a range of frequencies.

The denoising algorithm is a variant of the Wiener or optimal filtering
technique. The importance of the noise sampling is to get a noise
spectrum that will be divided out of the frequency spectrum of the
sample so it does not cut off frequencies above a certain level. I don't
know why playing with the sample level but I am not certain how the
filter is actually implemented nor the role that the sensitivity
parameter takes in the problem. (note everything is done in frequency
space, at least in the normal variant of the filter.)

there are a large number of discussions of the Wiener filter on the net,
but not being a dsp guru, I can't recommend one. One can get the chapter
from Numberical Recipes in pdf form plus a couple of Wikipedia articles.

Chuck

On 9/5/12 6:51 AM, rfwilmut wrote:

Quote:

If the noise level is, say, -40 dB and you increase it to, say, -20dB before sampling, then when you carry out the reduction process any wanted audio between -20 and -40 will be reduced in level on the assumption that it is noise, and the amount by which it is reduced will be different at different frequencies, so you are distorting wanted audio. The whole point of the sampling process is to determine below what level audio can be regarded as noise and therefore reduced in level, and to do so at each of a range of frequencies.

I tried to create a simple audio file to let me experiment with this issue, but what I got wasn't what I thought it was. I created a track 10 s worth of 400 Hz sine wave at 50%. I generated another track of white noise at 2%. I then merged the two tracks. Sure enough, it sounds like a nice tone with hiss overlaid, like a much-used cassette tape of a 400 Hz signal. I stuck a second's worth of 2% white noise on the front so I would have something to sample for suppressing noise.

I intended to try the trick of sampling the amplified noise before suppressing the noise in the whole file. But just to make sure I was on the right track I sampled the un-amplified noise and did the noise suppression with that. I expected the hiss (white noise) to vanish and something reasonably like a 400 Hz signal to remain.

Well, the whole wave form disappeared! Noise and sine wave were just gone, and there was nothing left.

Not surprising that I don't know how to interpret this result, since I don't know exactly what the Merge function does (although it appeared to do exactly what I wanted) and I don't know how AP's noise suppression works.

Any ideas what I did wrong, or suggestions on how I can create a simple file to experiment with this?

Thanks,
Wayne Durham
Blacksburg, Virginia

I just carried out the same exercise: created a mono track of 30 seconds of 1kHz tone, created a second mono track of 2% white noise, aligned the tracks to give 10 seconds of noise before the tone, merged, sampled, denoised: and got exactly what you would expect. Hiss removed, tone present as before and clean of hiss, with just a very slight 'puff' of filtered noise before the tone cut in. So what you did to get rid of the tone I don't know.

Just for interest I went back to the unsupressed version and increased the hiss level by 6dB, sampled that and carried out the denoising: no difference - but it's not a valid test with only one frequency. I would expect tampering with the noise level to have a damaging effect on a real audio signal, particularly low level sections. 'DeNoise' from Brian Davies has the same process, but you have more controls and can raise or lower the threshold: raising the threshold comes to exactly the same thing as amplifying the sample noise, and doing this progressively makes the audio more woolly.

I have published an article on digital noise reduction, with some examples, which may be of interest:

My problem appears to have been caused by the short duration (1 s) of noise in my file. [I meant to say "the one second of noise that preceded the 400 Hz signal been that had merged with noise." Subsequent references to "the noise" are to that one second piece I put there for sampling.] After getting the results I reported previously, I highlighted most of that noise and looked at its spectrum, and found that it had a large spike at 400 Hz. I guess the algorithm found the sample window too small and extended it to include some of the 400 Hz signal adjacent to it. Presumably the same thing happened when I sampled the noise [for the de-noising process], and it included 400 Hz in the sample. I tinkered with the settings and made the noisy part longer, and got the expected results.

Thanks for the link to your article. [Lots of good info, but several broken links.]

rfwilmut wrote:

Bull wrote:

I tried to create a simple audio file to let me experiment with this issue, but what I got wasn't what I thought it was. I created a track 10 s worth of 400 Hz sine wave at 50%. I generated another track of white noise at 2%. I then merged the two tracks. Sure enough, it sounds like a nice tone with hiss overlaid, like a much-used cassette tape of a 400 Hz signal. I stuck a second's worth of 2% white noise on the front so I would have something to sample for suppressing noise.

I intended to try the trick of sampling the amplified noise before suppressing the noise in the whole file. But just to make sure I was on the right track I sampled the un-amplified noise and did the noise suppression with that. I expected the hiss (white noise) to vanish and something reasonably like a 400 Hz signal to remain.

Well, the whole wave form disappeared! Noise and sine wave were just gone, and there was nothing left.

Not surprising that I don't know how to interpret this result, since I don't know exactly what the Merge function does (although it appeared to do exactly what I wanted) and I don't know how AP's noise suppression works.

Any ideas what I did wrong, or suggestions on how I can create a simple file to experiment with this?

Thanks,
Wayne Durham
Blacksburg, Virginia

I just carried out the same exercise: created a mono track of 30 seconds of 1kHz tone, created a second mono track of 2% white noise, aligned the tracks to give 10 seconds of noise before the tone, merged, sampled, denoised: and got exactly what you would expect. Hiss removed, tone present as before and clean of hiss, with just a very slight 'puff' of filtered noise before the tone cut in. So what you did to get rid of the tone I don't know.

Just for interest I went back to the unsupressed version and increased the hiss level by 6dB, sampled that and carried out the denoising: no difference - but it's not a valid test with only one frequency. I would expect tampering with the noise level to have a damaging effect on a real audio signal, particularly low level sections. 'DeNoise' from Brian Davies has the same process, but you have more controls and can raise or lower the threshold: raising the threshold comes to exactly the same thing as amplifying the sample noise, and doing this progressively makes the audio more woolly.

I have published an article on digital noise reduction, with some examples, which may be of interest:

Thank you for spotting the broken links. I've updated the ClickRepair and DeNoise links. Bias Soundsoap is no longer available (Bias have ceased business) and Dolby have apparently removed the 'Ken's Corner' article.

Some of the info is way out of my league but luckily a kind singer/song writer has agreed to help improve on the quality of the recordings i make...so your information will be very helpful.

I'm just not sure, after all that, whether the thinking was 'stick with the Ion to make the recordings' or buy an A/D box and use the bigger cd player (it sounds as though my plan to route the sound through the high quality cd player is not the way to go.)